Age | Commit message (Collapse) | Author |
|
The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself. This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box. The controlling user number should be made configurable.
JIRA ABE-2738
JIRA SWP-2846
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Factor out the equivalent function for analog.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines
Fix ISDN calling subaddr User Specified Odd/Even Flag
Calculation of the Odd/Even flag was wrong.
Implement correct algo, and set odd/even=0 if data would be truncated.
Only allow automatic calculation of the O/E flag, don't let dialplan influence.
(closes issue #19062)
Reported by: festr
Patches:
bug19062.diff2.txt uploaded by alecdavis (license 585)
Tested by: festr, alecdavis, rmudgett
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines
Crash if ISDN span layer 1 is down on initial load.
Regression from -r312575 B channel shifting during negotiation.
* Also combine updating the alarm flag with clearing the resetting flag.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
It was only used in a debug message and may not be correct anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
Merged revisions 312574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
Merged revisions 312573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
Issues with ISDN calls changing B channels during call negotiations.
The handling of the PROCEEDING message was not using the correct call
structure if the B channel was changed. (The same for PROGRESS.) The call
was also not hungup if the new B channel is not provisioned or is busy.
* Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
using the correct structure and B channel. If there is any problem with
the operations then the call is now hungup with an appropriate cause code.
* Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
correct structure by looking for the call and not using the channel ID.
NOTIFY is an exception with versions of libpri before v1.4.11 because a
call pointer is not available for Asterisk to use.
* Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
the correct structure by looking for the call and not using the channel
ID.
(closes issue #18313)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2620
(closes issue #18231)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2924
(closes issue #18488)
Reported by: jpokorny
JIRA SWP-2929
JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
JIRA DAHDI-406
JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines
Race condition when ISDN CallRerouting/CallDeflection invoked.
The queued AST_CONTROL_BUSY could sometimes be processed before the
call_forward dial string is recognized.
* Moved setting the call_forwarding dial string after sending a response
to the initiator and just queue an empty frame to wake up the media thread
instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond with an
error.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) | 1 line
Make pri parameter description consistent.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines
sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.
(closes issue #18874)
Reported by: cmaj
Patches:
patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)
JIRA SWP-3172
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting. Calls on hold or call-waiting
are not associated with any B channel.
JIRA LIBPRI-27
JIRA SWP-2547
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
No response sent for SIP CC subscribe/resubscribe request.
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.
The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.
JIRA SWP-2845
JIRA ABE-2736
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines
Merged revisions 305342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines
Merged revisions 305341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines
Obtain the pri lock for PRI queue counters.
Need to obtain the pri lock when calling pri_dump_info_str() to avoid a
reentrancy problem when calculating the Q.921 Q count statistic.
JIRA AST-484
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines
Add connected line chan_dahdi.conf pricpndialplan option.
* Added from_channel value to prilocaldialplan option.
JIRA ABE-2731
JIRA SWP-2842
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines
Merged revisions 303769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
Merged revisions 303765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing
through Asterisk. There is not enough information available at this point
to know if dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are not
adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early in
non-dial through extensions.
* Effectively revert issue #16789.
* Allow outgoing overlap dialing to hear dialtone and other early media.
A PROGRESS "inband-information is now available" message is now sent after
the SETUP_ACKNOWLEDGE message for non-digital calls. An
AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
messages for non-digital calls.
* Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
inconsistent with the cause codes.
* Added better protection from sending out of sequence messages by
combining several flags into a single enum value representing call
progress level.
* Added diagnostic messages for deferred overlap digits handling corner
cases.
(closes issue #17085)
Reported by: shawkris
(closes issue #18509)
Reported by: wimpy
Patches:
issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
and SS7 because of backporting requirements.
Tested by: wimpy, rmudgett
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines
Deadlock between dahdi_request() and pri_dchannel() processing an incomming call.
The sig_pri_new_ast_channel() is called with the channel private lock held
when pri_dchannel() calls it and no channel private lock held when
dahdi_request() calls it. The use of pri_grab() in
sig_pri_new_ast_channel() could leave the channel private lock held when
it returns if the lock was not held before calling it.
Make sig_pri_new_ast_channel() just lock the PRI span lock instead of
using pri_grab(). It is safe to do this because dahdi_request() does not
have the channel private lock and the deadlock potential with the PRI span
lock is only between pri_dchannel() and other threads.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines
Merged revision 300711 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines
A call retrieved from hold may wind up with no audio.
If the retrieved call is natively bridged then the call may not have any
audio path. The following warning message is given:
"Failed to add <dfd> to conference <chan>/<chan>: Invalid argument".
* Open the media on a B channel when pri_fixup_principle() moves the call
from a no_b_channel channel to a real channel.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
..........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
hold.
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines
Chan_dahdi sends an empty COLP on the bridged channel.
Chan_dahdi always inserts a connected party IE when you call from one
dahdi channel to another dahdi channel, even if no such information was
received on the 2nd channel. This clears the display of many phones.
* Removed leftover artifact from before the valid flag was added.
* Updated all of the channel's caller id information with the new
connected line information instead of just the string parts.
(closes issue #18508)
Reported by: wimpy
Patches:
issue18508_trunk.patch uploaded by rmudgett (license 664)
Tested by: wimpy, rmudgett
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Part of JIRA SWP-2687/ABE-2691.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r298195 | rmudgett | 2010-12-13 11:11:43 -0600 (Mon, 13 Dec 2010) | 33 lines
Merged revisions 298194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r298194 | rmudgett | 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) | 26 lines
Merged revisions 298193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines
Outgoing PRI/BRI calls cannot do DTMF triggered transfers.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING
message is not received. The debug output shows that the DTMF begin event
is seen, but the DTMF end event is missing. When the DTMF begin happens,
the call is muted so we now have one way audio (until a DTMF end event is
somehow seen).
* Made set the proceeding flag when the PRI_EVENT_ANSWER event is
received.
* Made absorb the DTMF begin and DTMF end events if we are overlap dialing
and have not seen a PROCEEDING message.
* Added a debug message when absorbing a DTMF event.
JIRA SWP-2690
JIRA ABE-2697
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r294823 | rmudgett | 2010-11-11 20:45:22 -0600 (Thu, 11 Nov 2010) | 25 lines
Merged revisions 294822 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines
Merged revisions 294821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines
Asterisk is getting a "No D-channels available!" warning message every 4 seconds.
Asterisk is just whining too much with this message: "No D-channels
available! Using Primary channel XXX as D-channel anyway!".
Filtered the message so it only comes out once if there is no D channel
available without an intervening D channel available period.
(closes issue #17270)
Reported by: jmls
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) | 17 lines
Analog lines do not transfer CONNECTED LINE or execute the interception macros.
Add connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code.
Note that if you create a three-way call in sig_analog before transferring
the call, the distinction of the caller/callee interception macros make
little sense. The interception macro writer needs to be prepared for
either caller/callee macro to be executed. The current implementation
swaps which caller/callee interception macro is executed after a three-way
call is created.
Review: https://reviewboard.asterisk.org/r/996/
JIRA ABE-2589
JIRA SWP-2372
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) | 1 line
No need to define the struct if there are no users.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) | 4 lines
Allow the DAHDI driver to compile, even with a sufficiently older version of libpri.
Fixes our Bamboo builds.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) | 2 lines
Several more defines that need to be altered for compiling against an older version of libpri
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) | 4 lines
Allow the DAHDI driver to compile, even with a sufficiently older version of libpri.
Fixes our Bamboo builds.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) | 19 lines
Connected line is not updated when chan_dahdi/sig_pri or chan_misdn transfers a call.
When a call is transfered by ECT or implicitly by disconnect in sig_pri or
implicitly by disconnect in chan_misdn, the connected line information is
not exchanged. The connected line interception macros also need to be
executed if defined.
The CALLER interception macro is executed for the held call.
The CALLEE interception macro is executed for the active/ringing call.
JIRA ABE-2589
JIRA SWP-2296
Patches:
abe_2589_c3bier.patch uploaded by rmudgett (license 664)
abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/958/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) | 7 lines
Send CONNECT_ACKNOWLEDGE for CIS calls too.
The originator of the Q.SIG call completion signaling link was not changed
to the active state when the CONNECT message came in. The T309 processing
would immediately kill the signaling link because it was not in the active
state.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) | 5 lines
Avoid deadlock processing incoming AOC-E messages.
Deadlock avoidance for the owner channel was not done when processing
incoming AOC-E messages.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r286118 | rmudgett | 2010-09-10 15:55:37 -0500 (Fri, 10 Sep 2010) | 25 lines
Merged revisions 286116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
Merged revisions 286113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) | 8 lines
Made output libpri event names if pri debugging is enabled when sig_pri processes them.
* Simplified CLI "pri debug xx span xx" command code and removed redundant
debugging enabled messages.
* Made CLI "pri debug xx span xx" command only close the debugging log
file if it was opened.
........
r284780 | rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines
Simplified pri_dchannel() poll timeout duration code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines
Merged revisions 283049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
Merged revisions 283048 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
Q931 - Sending PROGRESS after sending ALERTING is a protocol error
The PRI layer in chan_dadhi will check if a PROGRESS message has already
been sent, and not allow sending another (although that is technically
allowed by the Q931 spec), however it does not protect against sending an
ALERTING and then sending a PROGRESS message, which is a violation of the
specification.
Most switches don't seem to care too deeply about this, but some do, and
will disconnect the call when receiving this invalid sequence.
Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
A.5/Q.931 -- Overview protocol control (network side) point-point
(sheet 3 of 8)"
(closes issue #17874)
Reported by: nic_bellamy
Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines
Merged revisions 282607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines
Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
(closes issue #16770)
Reported by: jamicque
Patches:
20100413__issue16770.diff.txt uploaded by tilghman (license 14)
20100811__issue16770.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul 2010) | 7 lines
Fix compilation error in chan_dahdi (strdupa -> ast_strdupa).
(closes issue #17751)
Reported by: b11d
Patches:
strdupa_oops.diff uploaded by malcolmd (license 924)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines
Fix inband DTMF detection on outgoing ISDN calls.
This is a regression from the sig_pri split from chan_dahdi. When a call is
first initiated, the inband DTMF detector is not enabled if it's an outgoing
ISDN call. However, it needs to be turned on once the media path starts up.
This handling was put back in the open_media() callback of chan_dahdi. In
sig_pri, open_media() calls were added to a few places where it was needed,
including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and
PRI_EVENT_PROCEEDING.
Thanks to rmudgett for helping me with the patch!
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines
ss_thread calls pri_grab without lock during overlap dial
Recent changes to chan_dahdi with relation to overlap dialing call
pri_grab without first obtaining a lock.
(closes issue #17414)
Reported by: pdf
Patches:
bug17414.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn.
Review: https://reviewboard.asterisk.org/r/696/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
(closes issue #17401)
Reported by: avalentin
Patches:
issue17401_digit_timeout.patch uploaded by rmudgett (license 664)
Tested by: avalentin, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Add the ability to report waiting messages to ISDN endpoints (phones).
Relevant specification: EN 300 650 and EN 300 745
Review: https://reviewboard.asterisk.org/r/599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Add the ability to report malicious callers as an AMI event in the call
event class.
Relevant specification: EN 300 180
Review: https://reviewboard.asterisk.org/r/576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|